VAD refactoring: Code style changes of local function.

Changes applied to local GmmProbability():
* Replaced shift macros with shift operations
* Indentation and braces
* Removed redundant code
* Removed unnecessary type casts
* Name changes
* Adjusted comments

Tested with vad_unittests and audioproc_unittest

BUG=None
TEST=None

Review URL: https://webrtc-codereview.appspot.com/475002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1991 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
bjornv@webrtc.org 2012-04-05 10:22:02 +00:00
parent cac787842c
commit 2273f325b2

View File

@ -91,409 +91,385 @@ static const int16_t kLocalThresholdVAG[3] = { 94, 94, 94 };
static const int16_t kGlobalThresholdVAG[3] = { 1100, 1050, 1100 }; static const int16_t kGlobalThresholdVAG[3] = { 1100, 1050, 1100 };
// Calculates the probabilities for both speech and background noise using // Calculates the probabilities for both speech and background noise using
// Gaussian Mixture Models. A hypothesis-test is performed to decide which type // Gaussian Mixture Models (GMM). A hypothesis-test is performed to decide which
// of signal is most probable. // type of signal is most probable.
// //
// - inst [i/o] : Pointer to VAD instance // - self [i/o] : Pointer to VAD instance
// - feature_vector [i] : Feature vector = log10(energy in frequency band) // - feature_vector [i] : Feature vector = log10(energy in frequency band)
// - total_power [i] : Total power in audio frame. // - total_power [i] : Total power in audio frame.
// - frame_length [i] : Number of input samples // - frame_length [i] : Number of input samples
// //
// - returns : the VAD decision (0 - noise, 1 - speech). // - returns : the VAD decision (0 - noise, 1 - speech).
static int16_t GmmProbability(VadInstT *inst, int16_t *feature_vector, static int16_t GmmProbability(VadInstT* self, int16_t* feature_vector,
int16_t total_power, int frame_length) int16_t total_power, int frame_length) {
{ int n, k;
int n, k; int16_t backval;
int16_t backval; int16_t h0, h1;
int16_t h0, h1; int16_t ratvec, xval;
int16_t ratvec, xval; int16_t vadflag = 0;
int16_t vadflag; int16_t shifts0, shifts1;
int16_t shifts0, shifts1; int16_t tmp16, tmp16_1, tmp16_2;
int16_t tmp16, tmp16_1, tmp16_2; int16_t diff;
int16_t diff, nr, pos; int nr, pos;
int16_t nmk, nmk2, nmk3, smk, smk2, nsk, ssk; int16_t nmk, nmk2, nmk3, smk, smk2, nsk, ssk;
int16_t delt, ndelt; int16_t delt, ndelt;
int16_t maxspe, maxmu; int16_t maxspe, maxmu;
int16_t deltaN[kTableSize], deltaS[kTableSize]; int16_t deltaN[kTableSize], deltaS[kTableSize];
int16_t ngprvec[kTableSize], sgprvec[kTableSize]; int16_t ngprvec[kTableSize], sgprvec[kTableSize];
int32_t h0test, h1test; int32_t h0test, h1test;
int32_t tmp32_1, tmp32_2; int32_t tmp32_1, tmp32_2;
int32_t dotVal; int32_t dotVal;
int32_t nmid, smid; int32_t nmid, smid;
int32_t probn[kNumGaussians], probs[kNumGaussians]; int32_t probn[kNumGaussians], probs[kNumGaussians];
int16_t *nmean1ptr, *nmean2ptr, *smean1ptr, *smean2ptr, *nstd1ptr, *nstd2ptr, int16_t *nmean1ptr, *nmean2ptr, *smean1ptr, *smean2ptr;
*sstd1ptr, *sstd2ptr; int16_t *nstd1ptr, *nstd2ptr, *sstd1ptr, *sstd2ptr;
int16_t overhead1, overhead2, individualTest, totalTest; int16_t overhead1, overhead2, individualTest, totalTest;
// Set the thresholds to different values based on frame length // Set various thresholds based on frame lengths (80, 160 or 240 samples).
if (frame_length == 80) if (frame_length == 80) {
{ overhead1 = self->over_hang_max_1[0];
// 80 input samples overhead2 = self->over_hang_max_2[0];
overhead1 = inst->over_hang_max_1[0]; individualTest = self->individual[0];
overhead2 = inst->over_hang_max_2[0]; totalTest = self->total[0];
individualTest = inst->individual[0]; } else if (frame_length == 160) {
totalTest = inst->total[0]; overhead1 = self->over_hang_max_1[1];
} else if (frame_length == 160) overhead2 = self->over_hang_max_2[1];
{ individualTest = self->individual[1];
// 160 input samples totalTest = self->total[1];
overhead1 = inst->over_hang_max_1[1]; } else {
overhead2 = inst->over_hang_max_2[1]; overhead1 = self->over_hang_max_1[2];
individualTest = inst->individual[1]; overhead2 = self->over_hang_max_2[2];
totalTest = inst->total[1]; individualTest = self->individual[2];
} else totalTest = self->total[2];
{ }
// 240 input samples
overhead1 = inst->over_hang_max_1[2]; if (total_power > kMinEnergy) {
overhead2 = inst->over_hang_max_2[2]; // We have a signal present.
individualTest = inst->individual[2]; // Set pointers to the Gaussian parameters.
totalTest = inst->total[2]; nmean1ptr = &self->noise_means[0];
nmean2ptr = &self->noise_means[kNumChannels];
smean1ptr = &self->speech_means[0];
smean2ptr = &self->speech_means[kNumChannels];
nstd1ptr = &self->noise_stds[0];
nstd2ptr = &self->noise_stds[kNumChannels];
sstd1ptr = &self->speech_stds[0];
sstd2ptr = &self->speech_stds[kNumChannels];
dotVal = 0;
for (n = 0; n < kNumChannels; n++) {
// Perform for all channels.
pos = (n << 1);
xval = feature_vector[n];
// Probability for Noise, Q7 * Q20 = Q27.
tmp32_1 = WebRtcVad_GaussianProbability(xval, *nmean1ptr++, *nstd1ptr++,
&deltaN[pos]);
probn[0] = kNoiseDataWeights[n] * tmp32_1;
tmp32_1 = WebRtcVad_GaussianProbability(xval, *nmean2ptr++, *nstd2ptr++,
&deltaN[pos + 1]);
probn[1] = kNoiseDataWeights[n + kNumChannels] * tmp32_1;
h0test = probn[0] + probn[1]; // Q27
h0 = (int16_t) (h0test >> 12); // Q15
// Probability for Speech.
tmp32_1 = WebRtcVad_GaussianProbability(xval, *smean1ptr++, *sstd1ptr++,
&deltaS[pos]);
probs[0] = kSpeechDataWeights[n] * tmp32_1;
tmp32_1 = WebRtcVad_GaussianProbability(xval, *smean2ptr++, *sstd2ptr++,
&deltaS[pos + 1]);
probs[1] = kSpeechDataWeights[n + kNumChannels] * tmp32_1;
h1test = probs[0] + probs[1]; // Q27
h1 = (int16_t) (h1test >> 12); // Q15
// Calculate the log likelihood ratio. Approximate log2(H1/H0) with
// |shifts0| - |shifts1|.
shifts0 = WebRtcSpl_NormW32(h0test);
shifts1 = WebRtcSpl_NormW32(h1test);
if ((h0test > 0) && (h1test > 0)) {
ratvec = shifts0 - shifts1;
} else if (h1test > 0) {
ratvec = 31 - shifts1;
} else if (h0test > 0) {
ratvec = shifts0 - 31;
} else {
ratvec = 0;
}
// VAD decision with spectrum weighting.
dotVal += WEBRTC_SPL_MUL_16_16(ratvec, kSpectrumWeight[n]);
// Individual channel test.
if ((ratvec << 2) > individualTest) {
vadflag = 1;
}
// Probabilities used when updating model.
if (h0 > 0) {
tmp32_1 = probn[0] & 0xFFFFF000; // Q27
tmp32_2 = (tmp32_1 << 2); // Q29
ngprvec[pos] = (int16_t) WebRtcSpl_DivW32W16(tmp32_2, h0); // Q14
ngprvec[pos + 1] = 16384 - ngprvec[pos];
} else {
ngprvec[pos] = 16384;
ngprvec[pos + 1] = 0;
}
// Probabilities used when updating model.
if (h1 > 0) {
tmp32_1 = probs[0] & 0xFFFFF000;
tmp32_2 = (tmp32_1 << 2);
sgprvec[pos] = (int16_t) WebRtcSpl_DivW32W16(tmp32_2, h1);
sgprvec[pos + 1] = 16384 - sgprvec[pos];
} else {
sgprvec[pos] = 0;
sgprvec[pos + 1] = 0;
}
} }
if (total_power > kMinEnergy) // Overall test.
{ // If signal present at all if (dotVal >= totalTest) {
vadflag |= 1;
// Set pointers to the gaussian parameters
nmean1ptr = &inst->noise_means[0];
nmean2ptr = &inst->noise_means[kNumChannels];
smean1ptr = &inst->speech_means[0];
smean2ptr = &inst->speech_means[kNumChannels];
nstd1ptr = &inst->noise_stds[0];
nstd2ptr = &inst->noise_stds[kNumChannels];
sstd1ptr = &inst->speech_stds[0];
sstd2ptr = &inst->speech_stds[kNumChannels];
vadflag = 0;
dotVal = 0;
for (n = 0; n < kNumChannels; n++)
{ // For all channels
pos = WEBRTC_SPL_LSHIFT_W16(n, 1);
xval = feature_vector[n];
// Probability for Noise, Q7 * Q20 = Q27
tmp32_1 = WebRtcVad_GaussianProbability(xval, *nmean1ptr++, *nstd1ptr++,
&deltaN[pos]);
probn[0] = (int32_t)(kNoiseDataWeights[n] * tmp32_1);
tmp32_1 = WebRtcVad_GaussianProbability(xval, *nmean2ptr++, *nstd2ptr++,
&deltaN[pos + 1]);
probn[1] = (int32_t)(kNoiseDataWeights[n + kNumChannels] * tmp32_1);
h0test = probn[0] + probn[1]; // Q27
h0 = (int16_t)WEBRTC_SPL_RSHIFT_W32(h0test, 12); // Q15
// Probability for Speech
tmp32_1 = WebRtcVad_GaussianProbability(xval, *smean1ptr++, *sstd1ptr++,
&deltaS[pos]);
probs[0] = (int32_t)(kSpeechDataWeights[n] * tmp32_1);
tmp32_1 = WebRtcVad_GaussianProbability(xval, *smean2ptr++, *sstd2ptr++,
&deltaS[pos + 1]);
probs[1] = (int32_t)(kSpeechDataWeights[n + kNumChannels] * tmp32_1);
h1test = probs[0] + probs[1]; // Q27
h1 = (int16_t)WEBRTC_SPL_RSHIFT_W32(h1test, 12); // Q15
// Get likelihood ratio. Approximate log2(H1/H0) with shifts0 - shifts1
shifts0 = WebRtcSpl_NormW32(h0test);
shifts1 = WebRtcSpl_NormW32(h1test);
if ((h0test > 0) && (h1test > 0))
{
ratvec = shifts0 - shifts1;
} else if (h1test > 0)
{
ratvec = 31 - shifts1;
} else if (h0test > 0)
{
ratvec = shifts0 - 31;
} else
{
ratvec = 0;
}
// VAD decision with spectrum weighting
dotVal += WEBRTC_SPL_MUL_16_16(ratvec, kSpectrumWeight[n]);
// Individual channel test
if ((ratvec << 2) > individualTest)
{
vadflag = 1;
}
// Probabilities used when updating model
if (h0 > 0)
{
tmp32_1 = probn[0] & 0xFFFFF000; // Q27
tmp32_2 = WEBRTC_SPL_LSHIFT_W32(tmp32_1, 2); // Q29
ngprvec[pos] = (int16_t)WebRtcSpl_DivW32W16(tmp32_2, h0);
ngprvec[pos + 1] = 16384 - ngprvec[pos];
} else
{
ngprvec[pos] = 16384;
ngprvec[pos + 1] = 0;
}
// Probabilities used when updating model
if (h1 > 0)
{
tmp32_1 = probs[0] & 0xFFFFF000;
tmp32_2 = WEBRTC_SPL_LSHIFT_W32(tmp32_1, 2);
sgprvec[pos] = (int16_t)WebRtcSpl_DivW32W16(tmp32_2, h1);
sgprvec[pos + 1] = 16384 - sgprvec[pos];
} else
{
sgprvec[pos] = 0;
sgprvec[pos + 1] = 0;
}
}
// Overall test
if (dotVal >= totalTest)
{
vadflag |= 1;
}
// Set pointers to the means and standard deviations.
nmean1ptr = &inst->noise_means[0];
smean1ptr = &inst->speech_means[0];
nstd1ptr = &inst->noise_stds[0];
sstd1ptr = &inst->speech_stds[0];
maxspe = 12800;
// Update the model's parameters
for (n = 0; n < kNumChannels; n++)
{
pos = WEBRTC_SPL_LSHIFT_W16(n, 1);
// Get min value in past which is used for long term correction
backval = WebRtcVad_FindMinimum(inst, feature_vector[n], n); // Q4
// Compute the "global" mean, that is the sum of the two means weighted
nmid = WEBRTC_SPL_MUL_16_16(kNoiseDataWeights[n], *nmean1ptr); // Q7 * Q7
nmid += WEBRTC_SPL_MUL_16_16(kNoiseDataWeights[n+kNumChannels],
*(nmean1ptr+kNumChannels));
tmp16_1 = (int16_t)WEBRTC_SPL_RSHIFT_W32(nmid, 6); // Q8
for (k = 0; k < kNumGaussians; k++)
{
nr = pos + k;
nmean2ptr = nmean1ptr + k * kNumChannels;
smean2ptr = smean1ptr + k * kNumChannels;
nstd2ptr = nstd1ptr + k * kNumChannels;
sstd2ptr = sstd1ptr + k * kNumChannels;
nmk = *nmean2ptr;
smk = *smean2ptr;
nsk = *nstd2ptr;
ssk = *sstd2ptr;
// Update noise mean vector if the frame consists of noise only
nmk2 = nmk;
if (!vadflag)
{
// deltaN = (x-mu)/sigma^2
// ngprvec[k] = probn[k]/(probn[0] + probn[1])
delt = (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(ngprvec[nr],
deltaN[nr], 11); // Q14*Q11
nmk2 = nmk + (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(delt,
kNoiseUpdateConst,
22); // Q7+(Q14*Q15>>22)
}
// Long term correction of the noise mean
ndelt = WEBRTC_SPL_LSHIFT_W16(backval, 4);
ndelt -= tmp16_1; // Q8 - Q8
nmk3 = nmk2 + (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(ndelt,
kBackEta,
9); // Q7+(Q8*Q8)>>9
// Control that the noise mean does not drift to much
tmp16 = WEBRTC_SPL_LSHIFT_W16(k+5, 7);
if (nmk3 < tmp16)
nmk3 = tmp16;
tmp16 = WEBRTC_SPL_LSHIFT_W16(72+k-n, 7);
if (nmk3 > tmp16)
nmk3 = tmp16;
*nmean2ptr = nmk3;
if (vadflag)
{
// Update speech mean vector:
// deltaS = (x-mu)/sigma^2
// sgprvec[k] = probn[k]/(probn[0] + probn[1])
delt = (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(sgprvec[nr],
deltaS[nr],
11); // (Q14*Q11)>>11=Q14
tmp16 = (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(delt,
kSpeechUpdateConst,
21) + 1;
smk2 = smk + (tmp16 >> 1); // Q7 + (Q14 * Q15 >> 22)
// Control that the speech mean does not drift to much
maxmu = maxspe + 640;
if (smk2 < kMinimumMean[k])
smk2 = kMinimumMean[k];
if (smk2 > maxmu)
smk2 = maxmu;
*smean2ptr = smk2;
// (Q7>>3) = Q4
tmp16 = WEBRTC_SPL_RSHIFT_W16((smk + 4), 3);
tmp16 = feature_vector[n] - tmp16; // Q4
tmp32_1 = WEBRTC_SPL_MUL_16_16_RSFT(deltaS[nr], tmp16, 3);
tmp32_2 = tmp32_1 - (int32_t)4096; // Q12
tmp16 = WEBRTC_SPL_RSHIFT_W16((sgprvec[nr]), 2);
tmp32_1 = (int32_t)(tmp16 * tmp32_2);// (Q15>>3)*(Q14>>2)=Q12*Q12=Q24
tmp32_2 = WEBRTC_SPL_RSHIFT_W32(tmp32_1, 4); // Q20
// 0.1 * Q20 / Q7 = Q13
if (tmp32_2 > 0)
tmp16 = (int16_t)WebRtcSpl_DivW32W16(tmp32_2, ssk * 10);
else
{
tmp16 = (int16_t)WebRtcSpl_DivW32W16(-tmp32_2, ssk * 10);
tmp16 = -tmp16;
}
// divide by 4 giving an update factor of 0.025
tmp16 += 128; // Rounding
ssk += WEBRTC_SPL_RSHIFT_W16(tmp16, 8);
// Division with 8 plus Q7
if (ssk < kMinStd)
ssk = kMinStd;
*sstd2ptr = ssk;
} else
{
// Update GMM variance vectors
// deltaN * (feature_vector[n] - nmk) - 1, Q11 * Q4
tmp16 = feature_vector[n] - WEBRTC_SPL_RSHIFT_W16(nmk, 3);
// (Q15>>3) * (Q14>>2) = Q12 * Q12 = Q24
tmp32_1 = WEBRTC_SPL_MUL_16_16_RSFT(deltaN[nr], tmp16, 3) - 4096;
tmp16 = WEBRTC_SPL_RSHIFT_W16((ngprvec[nr]+2), 2);
tmp32_2 = (int32_t)(tmp16 * tmp32_1);
tmp32_1 = WEBRTC_SPL_RSHIFT_W32(tmp32_2, 14);
// Q20 * approx 0.001 (2^-10=0.0009766)
// Q20 / Q7 = Q13
tmp16 = (int16_t)WebRtcSpl_DivW32W16(tmp32_1, nsk);
if (tmp32_1 > 0)
tmp16 = (int16_t)WebRtcSpl_DivW32W16(tmp32_1, nsk);
else
{
tmp16 = (int16_t)WebRtcSpl_DivW32W16(-tmp32_1, nsk);
tmp16 = -tmp16;
}
tmp16 += 32; // Rounding
nsk += WEBRTC_SPL_RSHIFT_W16(tmp16, 6);
if (nsk < kMinStd)
nsk = kMinStd;
*nstd2ptr = nsk;
}
}
// Separate models if they are too close - nmid in Q14
nmid = WEBRTC_SPL_MUL_16_16(kNoiseDataWeights[n], *nmean1ptr);
nmid += WEBRTC_SPL_MUL_16_16(kNoiseDataWeights[n+kNumChannels], *nmean2ptr);
// smid in Q14
smid = WEBRTC_SPL_MUL_16_16(kSpeechDataWeights[n], *smean1ptr);
smid += WEBRTC_SPL_MUL_16_16(kSpeechDataWeights[n+kNumChannels], *smean2ptr);
// diff = "global" speech mean - "global" noise mean
diff = (int16_t)WEBRTC_SPL_RSHIFT_W32(smid, 9);
tmp16 = (int16_t)WEBRTC_SPL_RSHIFT_W32(nmid, 9);
diff -= tmp16;
if (diff < kMinimumDifference[n])
{
tmp16 = kMinimumDifference[n] - diff; // Q5
// tmp16_1 = ~0.8 * (kMinimumDifference - diff) in Q7
// tmp16_2 = ~0.2 * (kMinimumDifference - diff) in Q7
tmp16_1 = (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(13, tmp16, 2);
tmp16_2 = (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(3, tmp16, 2);
// First Gauss, speech model
tmp16 = tmp16_1 + *smean1ptr;
*smean1ptr = tmp16;
smid = WEBRTC_SPL_MUL_16_16(tmp16, kSpeechDataWeights[n]);
// Second Gauss, speech model
tmp16 = tmp16_1 + *smean2ptr;
*smean2ptr = tmp16;
smid += WEBRTC_SPL_MUL_16_16(tmp16, kSpeechDataWeights[n+kNumChannels]);
// First Gauss, noise model
tmp16 = *nmean1ptr - tmp16_2;
*nmean1ptr = tmp16;
nmid = WEBRTC_SPL_MUL_16_16(tmp16, kNoiseDataWeights[n]);
// Second Gauss, noise model
tmp16 = *nmean2ptr - tmp16_2;
*nmean2ptr = tmp16;
nmid += WEBRTC_SPL_MUL_16_16(tmp16, kNoiseDataWeights[n+kNumChannels]);
}
// Control that the speech & noise means do not drift to much
maxspe = kMaximumSpeech[n];
tmp16_2 = (int16_t)WEBRTC_SPL_RSHIFT_W32(smid, 7);
if (tmp16_2 > maxspe)
{ // Upper limit of speech model
tmp16_2 -= maxspe;
*smean1ptr -= tmp16_2;
*smean2ptr -= tmp16_2;
}
tmp16_2 = (int16_t)WEBRTC_SPL_RSHIFT_W32(nmid, 7);
if (tmp16_2 > kMaximumNoise[n])
{
tmp16_2 -= kMaximumNoise[n];
*nmean1ptr -= tmp16_2;
*nmean2ptr -= tmp16_2;
}
nmean1ptr++;
smean1ptr++;
nstd1ptr++;
sstd1ptr++;
}
inst->frame_counter++;
} else
{
vadflag = 0;
} }
// Hangover smoothing // Set pointers to the means and standard deviations.
if (!vadflag) nmean1ptr = &self->noise_means[0];
{ smean1ptr = &self->speech_means[0];
if (inst->over_hang > 0) nstd1ptr = &self->noise_stds[0];
{ sstd1ptr = &self->speech_stds[0];
vadflag = 2 + inst->over_hang;
inst->over_hang = inst->over_hang - 1; maxspe = 12800;
// Update the model parameters.
for (n = 0; n < kNumChannels; n++) {
pos = (n << 1);
// Get minimum value in past which is used for long term correction.
backval = WebRtcVad_FindMinimum(self, feature_vector[n], n); // Q4
// Compute the "global" mean, that is the sum of the two means weighted.
nmid = WEBRTC_SPL_MUL_16_16(kNoiseDataWeights[n], *nmean1ptr); // Q7 * Q7
nmid += WEBRTC_SPL_MUL_16_16(kNoiseDataWeights[n + kNumChannels],
*(nmean1ptr + kNumChannels));
tmp16_1 = (int16_t) (nmid >> 6); // Q8
for (k = 0; k < kNumGaussians; k++) {
nr = pos + k;
nmean2ptr = nmean1ptr + k * kNumChannels;
smean2ptr = smean1ptr + k * kNumChannels;
nstd2ptr = nstd1ptr + k * kNumChannels;
sstd2ptr = sstd1ptr + k * kNumChannels;
nmk = *nmean2ptr;
smk = *smean2ptr;
nsk = *nstd2ptr;
ssk = *sstd2ptr;
// Update noise mean vector if the frame consists of noise only.
nmk2 = nmk;
if (!vadflag) {
// deltaN = (x-mu)/sigma^2
// ngprvec[k] = probn[k]/(probn[0] + probn[1])
// (Q14 * Q11 >> 11) = Q14.
delt = (int16_t) WEBRTC_SPL_MUL_16_16_RSFT(ngprvec[nr], deltaN[nr],
11);
// Q7 + (Q14 * Q15 >> 22) = Q7.
nmk2 = nmk + (int16_t) WEBRTC_SPL_MUL_16_16_RSFT(delt,
kNoiseUpdateConst,
22);
} }
inst->num_of_speech = 0;
} else // Long term correction of the noise mean.
{ // Q8 - Q8 = Q8.
inst->num_of_speech = inst->num_of_speech + 1; ndelt = (backval << 4) - tmp16_1;
if (inst->num_of_speech > kMaxSpeechFrames) // Q7 + (Q8 * Q8) >> 9 = Q7.
{ nmk3 = nmk2 + (int16_t) WEBRTC_SPL_MUL_16_16_RSFT(ndelt, kBackEta, 9);
inst->num_of_speech = kMaxSpeechFrames;
inst->over_hang = overhead2; // Control that the noise mean does not drift to much.
} else tmp16 = (int16_t) ((k + 5) << 7);
inst->over_hang = overhead1; if (nmk3 < tmp16) {
nmk3 = tmp16;
}
tmp16 = (int16_t) ((72 + k - n) << 7);
if (nmk3 > tmp16) {
nmk3 = tmp16;
}
*nmean2ptr = nmk3;
if (vadflag) {
// Update speech mean vector:
// |deltaS| = (x-mu)/sigma^2
// sgprvec[k] = probn[k]/(probn[0] + probn[1])
// (Q14 * Q11) >> 11 = Q14.
delt = (int16_t) WEBRTC_SPL_MUL_16_16_RSFT(sgprvec[nr], deltaS[nr],
11);
// Q14 * Q15 >> 21 = Q8.
tmp16 = (int16_t) WEBRTC_SPL_MUL_16_16_RSFT(delt, kSpeechUpdateConst,
21);
// Q7 + (Q8 >> 1) = Q7. With rounding.
smk2 = smk + ((tmp16 + 1) >> 1);
// Control that the speech mean does not drift to much.
maxmu = maxspe + 640;
if (smk2 < kMinimumMean[k]) {
smk2 = kMinimumMean[k];
}
if (smk2 > maxmu) {
smk2 = maxmu;
}
*smean2ptr = smk2; // Q7.
// (Q7 >> 3) = Q4. With rounding.
tmp16 = ((smk + 4) >> 3);
tmp16 = feature_vector[n] - tmp16; // Q4
// (Q11 * Q4 >> 3) = Q12.
tmp32_1 = WEBRTC_SPL_MUL_16_16_RSFT(deltaS[nr], tmp16, 3);
tmp32_2 = tmp32_1 - 4096;
tmp16 = (sgprvec[nr] >> 2);
// (Q14 >> 2) * Q12 = Q24.
tmp32_1 = tmp16 * tmp32_2;
tmp32_2 = (tmp32_1 >> 4); // Q20
// 0.1 * Q20 / Q7 = Q13.
if (tmp32_2 > 0) {
tmp16 = (int16_t) WebRtcSpl_DivW32W16(tmp32_2, ssk * 10);
} else {
tmp16 = (int16_t) WebRtcSpl_DivW32W16(-tmp32_2, ssk * 10);
tmp16 = -tmp16;
}
// Divide by 4 giving an update factor of 0.025 (= 0.1 / 4).
// Note that division by 4 equals shift by 2, hence,
// (Q13 >> 8) = (Q13 >> 6) / 4 = Q7.
tmp16 += 128; // Rounding.
ssk += (tmp16 >> 8);
if (ssk < kMinStd) {
ssk = kMinStd;
}
*sstd2ptr = ssk;
} else {
// Update GMM variance vectors.
// deltaN * (feature_vector[n] - nmk) - 1
// Q4 - (Q7 >> 3) = Q4.
tmp16 = feature_vector[n] - (nmk >> 3);
// (Q11 * Q4 >> 3) = Q12.
tmp32_1 = WEBRTC_SPL_MUL_16_16_RSFT(deltaN[nr], tmp16, 3) - 4096;
// (Q14 >> 2) * Q12 = Q24.
tmp16 = ((ngprvec[nr] + 2) >> 2);
tmp32_2 = tmp16 * tmp32_1;
// Q20 * approx 0.001 (2^-10=0.0009766), hence,
// (Q24 >> 14) = (Q24 >> 4) / 2^10 = Q20.
tmp32_1 = (tmp32_2 >> 14);
// Q20 / Q7 = Q13.
if (tmp32_1 > 0) {
tmp16 = (int16_t) WebRtcSpl_DivW32W16(tmp32_1, nsk);
} else {
tmp16 = (int16_t) WebRtcSpl_DivW32W16(-tmp32_1, nsk);
tmp16 = -tmp16;
}
tmp16 += 32; // Rounding
nsk += (tmp16 >> 6); // Q13 >> 6 = Q7.
if (nsk < kMinStd) {
nsk = kMinStd;
}
*nstd2ptr = nsk;
}
}
// Separate models if they are too close.
// |nmid| in Q14 (= Q7 * Q7).
nmid = WEBRTC_SPL_MUL_16_16(kNoiseDataWeights[n], *nmean1ptr);
nmid += WEBRTC_SPL_MUL_16_16(kNoiseDataWeights[n + kNumChannels],
*nmean2ptr);
// |smid| in Q14 (= Q7 * Q7).
smid = WEBRTC_SPL_MUL_16_16(kSpeechDataWeights[n], *smean1ptr);
smid += WEBRTC_SPL_MUL_16_16(kSpeechDataWeights[n + kNumChannels],
*smean2ptr);
// |diff| = "global" speech mean - "global" noise mean.
// (Q14 >> 9) - (Q14 >> 9) = Q5.
diff = (int16_t) (smid >> 9) - (int16_t) (nmid >> 9);
if (diff < kMinimumDifference[n]) {
tmp16 = kMinimumDifference[n] - diff;
// |tmp16_1| = ~0.8 * (kMinimumDifference - diff) in Q7.
// |tmp16_2| = ~0.2 * (kMinimumDifference - diff) in Q7.
tmp16_1 = (int16_t) WEBRTC_SPL_MUL_16_16_RSFT(13, tmp16, 2);
tmp16_2 = (int16_t) WEBRTC_SPL_MUL_16_16_RSFT(3, tmp16, 2);
// First Gaussian, speech model.
tmp16 = tmp16_1 + *smean1ptr;
*smean1ptr = tmp16;
smid = WEBRTC_SPL_MUL_16_16(tmp16, kSpeechDataWeights[n]);
// Second Gaussian, speech model.
tmp16 = tmp16_1 + *smean2ptr;
*smean2ptr = tmp16;
smid += WEBRTC_SPL_MUL_16_16(tmp16,
kSpeechDataWeights[n + kNumChannels]);
// First Gaussian, noise model.
tmp16 = *nmean1ptr - tmp16_2;
*nmean1ptr = tmp16;
nmid = WEBRTC_SPL_MUL_16_16(tmp16, kNoiseDataWeights[n]);
// Second Gaussian, noise model.
tmp16 = *nmean2ptr - tmp16_2;
*nmean2ptr = tmp16;
nmid += WEBRTC_SPL_MUL_16_16(tmp16,
kNoiseDataWeights[n + kNumChannels]);
}
// Control that the speech & noise means do not drift to much.
maxspe = kMaximumSpeech[n];
tmp16_2 = (int16_t) (smid >> 7);
if (tmp16_2 > maxspe) {
// Upper limit of speech model.
tmp16_2 -= maxspe;
*smean1ptr -= tmp16_2;
*smean2ptr -= tmp16_2;
}
tmp16_2 = (int16_t) (nmid >> 7);
if (tmp16_2 > kMaximumNoise[n]) {
tmp16_2 -= kMaximumNoise[n];
*nmean1ptr -= tmp16_2;
*nmean2ptr -= tmp16_2;
}
nmean1ptr++;
smean1ptr++;
nstd1ptr++;
sstd1ptr++;
} }
return vadflag; self->frame_counter++;
}
// Smooth with respect to transition hysteresis.
if (!vadflag) {
if (self->over_hang > 0) {
vadflag = 2 + self->over_hang;
self->over_hang--;
}
self->num_of_speech = 0;
} else {
self->num_of_speech++;
if (self->num_of_speech > kMaxSpeechFrames) {
self->num_of_speech = kMaxSpeechFrames;
self->over_hang = overhead2;
} else {
self->over_hang = overhead1;
}
}
return vadflag;
} }
// Initialize the VAD. Set aggressiveness mode to default value. // Initialize the VAD. Set aggressiveness mode to default value.